PJSIP help for B2BUA config

Asterisk Asterisk SIP
1

Hi all,

i’m boring the community again with my configuration issue. I’m trying to let PJSIP wizard works as expected, in this context, where Asterisk is “Asterisk SBC” (obviously):

[OpenSIPS SIP router]----[Asterisk SBC]----[Provider SBC]

Here’s my pjsip.conf:

[global]
type = global

[transport-udp]
type=transport
protocol=udp    ;udp,tcp,tls,ws,wss,flow
bind=0.0.0.0

[trunk_defaults](!)
type = wizard
transport = ipv4
endpoint/allow_subscribe = no
endpoint/allow=!all,g729,g722,alaw
aor/qualify_frequency = 30
registration/expiration = 1800
has_hint = yes

and this my pjsip_wizard.conf:

[toip](trunk_defaults)
hint_context = from-toip
endpoint/context = from-toip
remote_hosts = [Provider SBC IP]
sends_registrations = no
accepts_registrations = no
sends_auth = no
accepts_auth = no

[voip](trunk_defaults)
hint_context = from-voip
endpoint/context = from-voip
remote_hosts = [OpenSIPS IP]
sends_registrations = no
accepts_registrations = no
sends_auth = no
accepts_auth = no

When i call from a SIP phone connected to OpenSIPS and the call correctly routed via Asterisk SBC, i got this on console:

[Mar  9 11:09:05] NOTICE[154365]: res_pjsip_session.c:3962 new_invite:  voip-trunk: Call (UDP:[OpenSIPS IP]:5060) to extension '12345' rejected because extension not found in context 'default'.
[Mar  9 11:09:05] NOTICE[154365]: res_pjsip_exten_state.c:418 new_subscribe: Endpoint 'voip-trunk' state subscription failed: Extension '12345' does not exist in context 'default' or has no associated hint

my extension.conf is as simple as possibile:

[general]
static=yes
writeprotect=no
;autofallthrough=no
;extenpatternmatchnew=no
clearglobalvars=no
userscontext=default

[globals]

[default]


[from-toip]
exten => _0X,1,Dial(SIP/voip-trunk/${EXTEN})

[from-voip]
exten => _0X,1,Dial(SIP/toip-trunk/${EXTEN})

What’s wrong? Thanks for help :slight_smile:

MP

2

You can’t specify trunk_defaults in pjsip.conf and have it inherit in pjsip_wizard.conf for one thing. As well you can inspect configuration from the CLI using “pjsip” CLI commands (use tab completion to explore) to see what Asterisk is actually configured to do and if it matches what you’ve specified.

3

Thanks,

i’v solved. I need to include “trunk_defaults” inside pjsip_wizard and fix transport=

Then i’ve changed extensions.conf with:

[from-voip]
exten => _0X.,1...

and now seems to work as expected!

4

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